1. Technical Field of the Invention
This invention relates to radio telecommunication systems and, more particularly, to a method of call control to minimize delays in launching multimedia or voice calls in a packet-switched radio telecommunications network.
2. Description of Related Art
In the methodology currently utilized for a call case involving two third generation IP-based mobile stations (MS1 and MS2), the mobile stations launch multicast discovery messages for either a Class-D H.323 gatekeeper at IP address 224.0.1.41 or a Session Initiation Protocol (SIP) proxy server at address 224.0.1.75. xe2x80x9cThird generationxe2x80x9d is the terminology used for launching multimedia or voice calls over packet-switched mobile access systems such as the General Packet Radio Service (GPRS) and the Universal Mobile Telecommunication System (UMTS), the wideband Code Division Multiple Access (CDMA) technique utilized on GPRS. In theory, any gatekeeper/server that joins the above mentioned Class-D multicast group is free to reply to the discovery message. Once an MS-to-MS third generation call is launched, the path traversed by the media stream can be very random, and could involve multiple Gateway GPRS Service Nodes (GGSNs) and Internet Service Providers (ISPs)/Public Internet Protocol (IP) networks. The delay induced in the core network and the ISPs could vary from 10 ms to hundreds of milliseconds depending on the actual path taken by the Packet Data Unit (PDU).
Recommendations from the International Telecommunications Union-Telecommunications Standardization Sector (ITU-T), in particular ITU-T Recommendations G.114 and G.131, specify the effect of various delay ranges on the quality of a voice call. These recommendations state that any call taking over 400 ms for end-to-end delivery of the voice payload is considered unacceptable for normal interactive communication, and half-duplex procedures are then required. Given the sizable delay, already accumulated in packetizing and sending the PDU over the air interface, it is imperative that the core network create the shortest additional delay possible in order to minimize the overall delay in end-to-end delivery of the voice payload.
However, under existing standards, a calling MS""s GGSN will route a voice or multimedia call to the MS""s ISP where the MS-to-MS call is diverted into the public Internet. This leads to inordinate delays and the actual expiration of call control timers. Therefore, it would be advantageous to have a method of call control that minimizes delays in launching multimedia or voice calls over packet-switched mobile access systems. The present invention provides such a method.
The present invention is a method of call control in a packet-switched radio telecommunication network that minimizes delays in launching a voice call from a first Internet Protocol (IP)-based mobile station (MS) to a second IP-based MS. The method includes the steps of preventing voice traffic from being routed to an Internet Service Provider (ISP), and setting up an optimized path for voice traffic from the first MS to the second MS. The optimized path may be set up by creating a shortest route tunnel between a first service node serving the first MS and a second service node serving the second MS. The tunnel may be established between a first Serving GPRS service node (SGSN1) serving the first MS and a second SGSN (SGSN2) serving the second MS. Alternatively, the tunnel may be established between the base station controllers (BSCs) of each MS""s serving radio base station in the case of UMTS or CDMA2000 where packet-based inter-BSC links have been defined.